An increasing number of devices conforming to the aforesaid standards are available on the market. Support of these standards, however, is not sufficient to implement essential performance features of a “multi-room audio system”. These performance features are:                Synchronous playback of the same audio data or of different channels of the same audio source by different audio adapters.        Synchronous start of an additional audio adapter during playback.        
A trivial approach would be a remote-control start of the autonomous playback station by means of a remote-control signal and as synchronously as possible. This approach, however, results in a signal offset in the range of 200-500 msec. This would not be practical and the corresponding approach is therefore not suitable for the implementation of a multi-room audio system.
The long response time is due to the differently sized pre-buffers of the audio adapters, which are filled in accordance with the audio format used once the playback function is activated but before the playback process starts.
In addition, TCP/IP is a packet-oriented standard, where, in principle, no statements relating to the transfer times between the transmission and processing of information can be made.
Besides the response time problem described above, there is the problem of unsynchronised phases. The playback rate of each audio adapter is, as a rule, dependent on the quartz of the relevant converter chip. The tolerances of each individual quartz result in that the phases of the audio adapters will increasingly differ from each other over time (drift). Both phenomena add up, resulting in a signal offset and unsynchronous playback in the current state of the art.
Existing multi-room audio systems implementing the aforesaid performance features, i.e. synchronous playback or synchronous additional start, are based on proprietary protocols.
As an alternative to TCP, there is the RTP transmission standard (RTP=Real-Time Transport Protocol) for the synchronised playback of media data (video+audio). However, the use of RTP for the implementation of a multi-room audio system has the following drawbacks:                The resolution of the synchronisation information is too low for audio data.        RTP is based on UDP. On the internet and also in LAN environments, however, http based on tcp is much more common and must be regarded as a de-facto standard.        UDP is a “connectionless” transmission standard, which, in contrast to tcp, does not guarantee transmission. This means, data packets may be lost during transmission. This is acceptable provided that the main area of use of RTP is video and not hi-fi audio. In case of video, such an incident (“packet loss”) will result in a frame drop, which, in case of doubt, will not be noticed. In case of audio, however, the effect is a clearly audible gap. UDP packet loss is, in particular, a problem in IEEE 802.11 wireless networks.        
While time or clock information is partly provided in computer networks, said information is, in particular, transmitted in the data link layer of the OSI layer model (OSI=Open Systems Interconnection). As a consequence, said information depends on the hardware and is not suitable for the purpose of synchronising audio playback devices within a home network. In addition, heterogeneous networks including different transmission technologies (such as Ethernet, Wifi, Bluetooth, Powerline, . . . ) are increasingly used, in particular in the home sector, so that the use of said technology-specific information limits the applicability of the invention.
It is therefore an object of the present invention to provide a method and an arrangement for synchronising data streams in networks and a corresponding computer program and corresponding computer-readable storage medium which avoid the aforesaid drawbacks and, in particular, allow data streams that do not comprise time information to be played back synchronously.